[ntp:questions] Accuracy of audio tones via VOIP

Robert Scott no-one at notreal.invalid
Tue Jul 9 14:35:02 UTC 2013


The NIST has a telephone service that carries the audio of WWV.  When
this service is accessed by a landline or a cellphone the playback of
the audio is tightly locked to the phone network system clock.
Therefore the user can count on fairly accurate audio frequency
rendering of the NIST standard audio frequency tones.

But what about accessing this NIST phone service via Skype and other
VOIP technologies?  In these technologies it is impossible to tightly
lock the playback to the system clock because of the indeterminacy of
internet delays.  So how does Skype on my desktop computer work?  It
can use the local oscillators (system or soundcard) to manage
playback, but that is not guaranteed to be so good.  Over the long
haul it is possible for Skype to lock its playback speed to the source
clock, but that is a slow process, as SNTP applications show.  Over a
short 3-minute phone call there is very little trimming of the
playback rate that is possible using only the arrival timing of the
internet packets.  So I have to wonder if the playback frequency of
the NIST tones over the telephone via VOIP are potentially in error by
as much as the local quartz oscillator frequencies?  Does anyone have
any insight on how Skype and other VOIP systems manage record/playback
rate synchronization?



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